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1 /* | 1 /* |
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ | 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ |
12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ | 12 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
16 #include "webrtc/base/buffer.h" | 16 #include "webrtc/base/buffer.h" |
17 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" | 17 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" |
18 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" | 18 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h" |
19 | 19 |
20 namespace webrtc { | 20 namespace webrtc { |
21 | 21 |
22 struct CodecInst; | 22 struct CodecInst; |
23 | 23 |
24 class AudioEncoderG722 final : public AudioEncoder { | 24 class AudioEncoderG722 final : public AudioEncoder { |
25 public: | 25 public: |
26 using AudioEncoder::EncodeInternal; | |
27 | |
28 struct Config { | 26 struct Config { |
29 bool IsOk() const; | 27 bool IsOk() const; |
30 | 28 |
31 int payload_type = 9; | 29 int payload_type = 9; |
32 int frame_size_ms = 20; | 30 int frame_size_ms = 20; |
33 size_t num_channels = 1; | 31 size_t num_channels = 1; |
34 }; | 32 }; |
35 | 33 |
36 explicit AudioEncoderG722(const Config& config); | 34 explicit AudioEncoderG722(const Config& config); |
37 explicit AudioEncoderG722(const CodecInst& codec_inst); | 35 explicit AudioEncoderG722(const CodecInst& codec_inst); |
38 ~AudioEncoderG722() override; | 36 ~AudioEncoderG722() override; |
39 | 37 |
40 size_t MaxEncodedBytes() const override; | 38 size_t MaxEncodedBytes() const override; |
41 int SampleRateHz() const override; | 39 int SampleRateHz() const override; |
42 size_t NumChannels() const override; | 40 size_t NumChannels() const override; |
43 int RtpTimestampRateHz() const override; | 41 int RtpTimestampRateHz() const override; |
44 size_t Num10MsFramesInNextPacket() const override; | 42 size_t Num10MsFramesInNextPacket() const override; |
45 size_t Max10MsFramesInAPacket() const override; | 43 size_t Max10MsFramesInAPacket() const override; |
46 int GetTargetBitrate() const override; | 44 int GetTargetBitrate() const override; |
47 void Reset() override; | 45 void Reset() override; |
48 | 46 |
49 protected: | 47 protected: |
50 EncodedInfo EncodeInternal(uint32_t rtp_timestamp, | 48 EncodedInfo EncodeImpl(uint32_t rtp_timestamp, |
51 rtc::ArrayView<const int16_t> audio, | 49 rtc::ArrayView<const int16_t> audio, |
52 rtc::Buffer* encoded) override; | 50 rtc::Buffer* encoded) override; |
53 | 51 |
54 private: | 52 private: |
55 // The encoder state for one channel. | 53 // The encoder state for one channel. |
56 struct EncoderState { | 54 struct EncoderState { |
57 G722EncInst* encoder; | 55 G722EncInst* encoder; |
58 std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding. | 56 std::unique_ptr<int16_t[]> speech_buffer; // Queued up for encoding. |
59 rtc::Buffer encoded_buffer; // Already encoded. | 57 rtc::Buffer encoded_buffer; // Already encoded. |
60 EncoderState(); | 58 EncoderState(); |
61 ~EncoderState(); | 59 ~EncoderState(); |
62 }; | 60 }; |
63 | 61 |
64 size_t SamplesPerChannel() const; | 62 size_t SamplesPerChannel() const; |
65 | 63 |
66 const size_t num_channels_; | 64 const size_t num_channels_; |
67 const int payload_type_; | 65 const int payload_type_; |
68 const size_t num_10ms_frames_per_packet_; | 66 const size_t num_10ms_frames_per_packet_; |
69 size_t num_10ms_frames_buffered_; | 67 size_t num_10ms_frames_buffered_; |
70 uint32_t first_timestamp_in_buffer_; | 68 uint32_t first_timestamp_in_buffer_; |
71 const std::unique_ptr<EncoderState[]> encoders_; | 69 const std::unique_ptr<EncoderState[]> encoders_; |
72 rtc::Buffer interleave_buffer_; | 70 rtc::Buffer interleave_buffer_; |
73 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722); | 71 RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722); |
74 }; | 72 }; |
75 | 73 |
76 } // namespace webrtc | 74 } // namespace webrtc |
77 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ | 75 #endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_ |
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